ilabs-flir 2.0.3 → 2.0.4

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/PassiveServerMediaSubsession.hh

@@ -1,83 +0,0 @@
-/**********
-This library is free software; you can redistribute it and/or modify it under
-the terms of the GNU Lesser General Public License as published by the
-Free Software Foundation; either version 3 of the License, or (at your
-option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
-
-This library is distributed in the hope that it will be useful, but WITHOUT
-ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
-FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
-more details.
-
-You should have received a copy of the GNU Lesser General Public License
-along with this library; if not, write to the Free Software Foundation, Inc.,
-51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
-**********/
-// "liveMedia"
-// Copyright (c) 1996-2023 Live Networks, Inc.  All rights reserved.
-// A 'ServerMediaSubsession' object that represents an existing
-// 'RTPSink', rather than one that creates new 'RTPSink's on demand.
-// C++ header
-
-#ifndef _PASSIVE_SERVER_MEDIA_SUBSESSION_HH
-#define _PASSIVE_SERVER_MEDIA_SUBSESSION_HH
-
-#ifndef _SERVER_MEDIA_SESSION_HH
-#include "ServerMediaSession.hh"
-#endif
-
-#ifndef _RTP_SINK_HH
-#include "RTPSink.hh"
-#endif
-#ifndef _RTCP_HH
-#include "RTCP.hh"
-#endif
-
-class LIVEMEDIA_API PassiveServerMediaSubsession: public ServerMediaSubsession {
-public:
-  static PassiveServerMediaSubsession* createNew(RTPSink& rtpSink,
-						 RTCPInstance* rtcpInstance = NULL);
-
-protected:
-  PassiveServerMediaSubsession(RTPSink& rtpSink, RTCPInstance* rtcpInstance);
-      // called only by createNew();
-  virtual ~PassiveServerMediaSubsession();
-
-  virtual Boolean rtcpIsMuxed();
-
-protected: // redefined virtual functions
-  virtual char const* sdpLines(int addressFamily);
-  virtual void getStreamParameters(unsigned clientSessionId,
-				   struct sockaddr_storage const& clientAddress,
-                                   Port const& clientRTPPort,
-                                   Port const& clientRTCPPort,
-				   int tcpSocketNum,
-                                   unsigned char rtpChannelId,
-                                   unsigned char rtcpChannelId,
-				   TLSState* tlsSTate,
-                                   struct sockaddr_storage& destinationAddress,
-				   u_int8_t& destinationTTL,
-                                   Boolean& isMulticast,
-                                   Port& serverRTPPort,
-                                   Port& serverRTCPPort,
-                                   void*& streamToken);
-  virtual void startStream(unsigned clientSessionId, void* streamToken,
-			   TaskFunc* rtcpRRHandler,
-			   void* rtcpRRHandlerClientData,
-                           unsigned short& rtpSeqNum,
-                           unsigned& rtpTimestamp,
-			   ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
-                           void* serverRequestAlternativeByteHandlerClientData);
-  virtual float getCurrentNPT(void* streamToken);
-  virtual void getRTPSinkandRTCP(void* streamToken,
-				 RTPSink const*& rtpSink, RTCPInstance const*& rtcp);
-  virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
-
-protected:
-  char* fSDPLines;
-  RTPSink& fRTPSink;
-  RTCPInstance* fRTCPInstance;
-  HashTable* fClientRTCPSourceRecords; // indexed by client session id; used to implement RTCP "RR" handling
-};
-
-#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/ProxyServerMediaSession.hh

@@ -1,238 +0,0 @@
-/**********
-This library is free software; you can redistribute it and/or modify it under
-the terms of the GNU Lesser General Public License as published by the
-Free Software Foundation; either version 3 of the License, or (at your
-option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
-
-This library is distributed in the hope that it will be useful, but WITHOUT
-ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
-FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
-more details.
-
-You should have received a copy of the GNU Lesser General Public License
-along with this library; if not, write to the Free Software Foundation, Inc.,
-51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
-**********/
-// "liveMedia"
-// Copyright (c) 1996-2023 Live Networks, Inc.  All rights reserved.
-// A subclass of "ServerMediaSession" that can be used to create a (unicast) RTSP servers that acts as a 'proxy' for
-// another (unicast or multicast) RTSP/RTP stream.
-// C++ header
-
-#ifndef _PROXY_SERVER_MEDIA_SESSION_HH
-#define _PROXY_SERVER_MEDIA_SESSION_HH
-
-#ifndef _SERVER_MEDIA_SESSION_HH
-#include "ServerMediaSession.hh"
-#endif
-#ifndef _MEDIA_SESSION_HH
-#include "MediaSession.hh"
-#endif
-#ifndef _RTSP_CLIENT_HH
-#include "RTSPClient.hh"
-#endif
-#ifndef _MEDIA_TRANSCODING_TABLE_HH
-#include "MediaTranscodingTable.hh"
-#endif
-
-// A subclass of "RTSPClient", used to refer to the particular "ProxyServerMediaSession" object being used.
-// It is used only within the implementation of "ProxyServerMediaSession", but is defined here, in case developers wish to
-// subclass it.
-
-class LIVEMEDIA_API ProxyRTSPClient: public RTSPClient {
-public:
-  ProxyRTSPClient(class ProxyServerMediaSession& ourServerMediaSession, char const* rtspURL,
-                  char const* username, char const* password,
-                  portNumBits tunnelOverHTTPPortNum, int verbosityLevel, int socketNumToServer);
-  virtual ~ProxyRTSPClient();
-
-  void continueAfterDESCRIBE(char const* sdpDescription);
-  void continueAfterLivenessCommand(int resultCode, Boolean serverSupportsGetParameter);
-  void continueAfterSETUP(int resultCode);
-  void continueAfterPLAY(int resultCode);
-  void scheduleReset();
-
-private:
-  void reset();
-  int connectToServer(int socketNum, portNumBits remotePortNum);
-
-  Authenticator* auth() { return fOurAuthenticator; }
-
-  void scheduleLivenessCommand();
-  static void sendLivenessCommand(void* clientData);
-  void doReset();
-  static void doReset(void* clientData);
-
-  void scheduleDESCRIBECommand();
-  static void sendDESCRIBE(void* clientData);
-  void sendDESCRIBE();
-
-  static void subsessionTimeout(void* clientData);
-  void handleSubsessionTimeout();
-
-private:
-  friend class ProxyServerMediaSession;
-  friend class ProxyServerMediaSubsession;
-  ProxyServerMediaSession& fOurServerMediaSession;
-  char* fOurURL;
-  Authenticator* fOurAuthenticator;
-  Boolean fStreamRTPOverTCP;
-  class ProxyServerMediaSubsession *fSetupQueueHead, *fSetupQueueTail;
-  unsigned fNumSetupsDone;
-  unsigned fNextDESCRIBEDelay; // in seconds
-  Boolean fServerSupportsGetParameter, fLastCommandWasPLAY, fDoneDESCRIBE;
-  TaskToken fLivenessCommandTask, fDESCRIBECommandTask, fSubsessionTimerTask, fResetTask;
-};
-
-
-typedef ProxyRTSPClient*
-createNewProxyRTSPClientFunc(ProxyServerMediaSession& ourServerMediaSession,
-			     char const* rtspURL,
-			     char const* username, char const* password,
-			     portNumBits tunnelOverHTTPPortNum, int verbosityLevel,
-			     int socketNumToServer);
-ProxyRTSPClient*
-defaultCreateNewProxyRTSPClientFunc(ProxyServerMediaSession& ourServerMediaSession,
-				    char const* rtspURL,
-				    char const* username, char const* password,
-				    portNumBits tunnelOverHTTPPortNum, int verbosityLevel,
-				    int socketNumToServer);
-
-class LIVEMEDIA_API ProxyServerMediaSession: public ServerMediaSession {
-public:
-  static ProxyServerMediaSession* createNew(UsageEnvironment& env,
-					    GenericMediaServer* ourMediaServer, // Note: We can be used by just one server
-					    char const* inputStreamURL, // the "rtsp://" URL of the stream we'll be proxying
-					    char const* streamName = NULL,
-					    char const* username = NULL, char const* password = NULL,
-					    portNumBits tunnelOverHTTPPortNum = 0,
-					        // for streaming the *proxied* (i.e., back-end) stream
-					    int verbosityLevel = 0,
-					    int socketNumToServer = -1,
-					    MediaTranscodingTable* transcodingTable = NULL);
-      // Hack: "tunnelOverHTTPPortNum" == 0xFFFF (i.e., all-ones) means: Stream RTP/RTCP-over-TCP, but *not* using HTTP
-      // "verbosityLevel" == 1 means display basic proxy setup info; "verbosityLevel" == 2 means display RTSP client protocol also.
-      // If "socketNumToServer" is >= 0, then it is the socket number of an already-existing TCP connection to the server.
-      //      (In this case, "inputStreamURL" must point to the socket's endpoint, so that it can be accessed via the socket.)
-
-  virtual ~ProxyServerMediaSession();
-
-  char const* url() const;
-
-  char describeCompletedFlag;
-    // initialized to 0; set to 1 when the back-end "DESCRIBE" completes.
-    // (This can be used as a 'watch variable' in "doEventLoop()".)
-  Boolean describeCompletedSuccessfully() const { return fClientMediaSession != NULL; }
-    // This can be used - along with "describeCompletedFlag" - to check whether the back-end "DESCRIBE" completed *successfully*.
-
-protected:
-  ProxyServerMediaSession(UsageEnvironment& env, GenericMediaServer* ourMediaServer,
-			  char const* inputStreamURL, char const* streamName,
-			  char const* username, char const* password,
-			  portNumBits tunnelOverHTTPPortNum, int verbosityLevel,
-			  int socketNumToServer,
-			  MediaTranscodingTable* transcodingTable,
-			  createNewProxyRTSPClientFunc* ourCreateNewProxyRTSPClientFunc
-			  = defaultCreateNewProxyRTSPClientFunc,
-			  portNumBits initialPortNum = 6970,
-			  Boolean multiplexRTCPWithRTP = False);
-
-  // If you subclass "ProxyRTSPClient", then you will also need to define your own function
-  // - with signature "createNewProxyRTSPClientFunc" (see above) - that creates a new object
-  // of this subclass.  You should also subclass "ProxyServerMediaSession" and, in your
-  // subclass's constructor, initialize the parent class (i.e., "ProxyServerMediaSession")
-  // constructor by passing your new function as the "ourCreateNewProxyRTSPClientFunc"
-  // parameter.
-
-  // Subclasses may redefine the following functions, if they want "ProxyServerSubsession"s
-  // to create subclassed "Groupsock" and/or "RTCPInstance" objects:
-  virtual Groupsock* createGroupsock(struct sockaddr_storage const& addr, Port port);
-  virtual RTCPInstance* createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */
-				   unsigned char const* cname, RTPSink* sink);
-
-  virtual Boolean allowProxyingForSubsession(MediaSubsession const& mss);
-  // By default, this function always returns True.  However, a subclass may redefine this
-  // if it wishes to restrict which subsessions of a stream get proxied - e.g., if it wishes
-  // to proxy only video tracks, but not audio (or other) tracks.
-
-protected:
-  GenericMediaServer* fOurMediaServer;
-  ProxyRTSPClient* fProxyRTSPClient;
-  MediaSession* fClientMediaSession;
-
-private:
-  friend class ProxyRTSPClient;
-  friend class ProxyServerMediaSubsession;
-  void continueAfterDESCRIBE(char const* sdpDescription);
-  void resetDESCRIBEState(); // undoes what was done by "contineAfterDESCRIBE()"
-
-private:
-  int fVerbosityLevel;
-  class PresentationTimeSessionNormalizer* fPresentationTimeSessionNormalizer;
-  createNewProxyRTSPClientFunc* fCreateNewProxyRTSPClientFunc;
-  MediaTranscodingTable* fTranscodingTable;
-  portNumBits fInitialPortNum;
-  Boolean fMultiplexRTCPWithRTP;
-};
-
-
-////////// PresentationTimeSessionNormalizer and PresentationTimeSubsessionNormalizer definitions //////////
-
-// The following two classes are used by proxies to convert incoming streams' presentation times into wall-clock-aligned
-// presentation times that are suitable for our "RTPSink"s (for the corresponding outgoing streams).
-// (For multi-subsession (i.e., audio+video) sessions, the outgoing streams' presentation times retain the same relative
-//  separation as those of the incoming streams.)
-
-class LIVEMEDIA_API PresentationTimeSubsessionNormalizer: public FramedFilter {
-public:
-  void setRTPSink(RTPSink* rtpSink) { fRTPSink = rtpSink; }
-
-private:
-  friend class PresentationTimeSessionNormalizer;
-  PresentationTimeSubsessionNormalizer(PresentationTimeSessionNormalizer& parent, FramedSource* inputSource, RTPSource* rtpSource,
-				       char const* codecName, PresentationTimeSubsessionNormalizer* next);
-      // called only from within "PresentationTimeSessionNormalizer"
-  virtual ~PresentationTimeSubsessionNormalizer();
-
-  static void afterGettingFrame(void* clientData, unsigned frameSize,
-                                unsigned numTruncatedBytes,
-                                struct timeval presentationTime,
-                                unsigned durationInMicroseconds);
-  void afterGettingFrame(unsigned frameSize,
-			 unsigned numTruncatedBytes,
-			 struct timeval presentationTime,
-			 unsigned durationInMicroseconds);
-
-private: // redefined virtual functions:
-  virtual void doGetNextFrame();
-
-private:
-  PresentationTimeSessionNormalizer& fParent;
-  RTPSource* fRTPSource;
-  RTPSink* fRTPSink;
-  char const* fCodecName;
-  PresentationTimeSubsessionNormalizer* fNext;
-};
-
-class LIVEMEDIA_API PresentationTimeSessionNormalizer: public Medium {
-public:
-  PresentationTimeSessionNormalizer(UsageEnvironment& env);
-  virtual ~PresentationTimeSessionNormalizer();
-
-  PresentationTimeSubsessionNormalizer*
-  createNewPresentationTimeSubsessionNormalizer(FramedSource* inputSource, RTPSource* rtpSource, char const* codecName);
-
-private: // called only from within "~PresentationTimeSubsessionNormalizer":
-  friend class PresentationTimeSubsessionNormalizer;
-  void normalizePresentationTime(PresentationTimeSubsessionNormalizer* ssNormalizer,
-				 struct timeval& toPT, struct timeval const& fromPT);
-  void removePresentationTimeSubsessionNormalizer(PresentationTimeSubsessionNormalizer* ssNormalizer);
-
-private:
-  PresentationTimeSubsessionNormalizer* fSubsessionNormalizers;
-  PresentationTimeSubsessionNormalizer* fMasterSSNormalizer; // used for subsessions that have been RTCP-synced
-
-  struct timeval fPTAdjustment; // Added to (RTCP-synced) subsession presentation times to 'normalize' them with wall-clock time.
-};
-
-#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/QCELPAudioRTPSource.hh

@@ -1,39 +0,0 @@
-/**********
-This library is free software; you can redistribute it and/or modify it under
-the terms of the GNU Lesser General Public License as published by the
-Free Software Foundation; either version 3 of the License, or (at your
-option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
-
-This library is distributed in the hope that it will be useful, but WITHOUT
-ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
-FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
-more details.
-
-You should have received a copy of the GNU Lesser General Public License
-along with this library; if not, write to the Free Software Foundation, Inc.,
-51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
-**********/
-// "liveMedia"
-// Copyright (c) 1996-2023 Live Networks, Inc.  All rights reserved.
-// Qualcomm "PureVoice" (aka. "QCELP") Audio RTP Sources
-// C++ header
-
-#ifndef _QCELP_AUDIO_RTP_SOURCE_HH
-#define _QCELP_AUDIO_RTP_SOURCE_HH
-
-#ifndef _RTP_SOURCE_HH
-#include "RTPSource.hh"
-#endif
-
-class LIVEMEDIA_API QCELPAudioRTPSource {
-public:
-  static FramedSource* createNew(UsageEnvironment& env,
-				 Groupsock* RTPgs,
-				 RTPSource*& resultRTPSource,
-				 unsigned char rtpPayloadFormat = 12,
-				 unsigned rtpTimestampFrequency = 8000);
-      // This returns a source to read from, but "resultRTPSource" will
-      // point to RTP-related state.
-};
-
-#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/QuickTimeFileSink.hh

@@ -1,192 +0,0 @@
-/**********
-This library is free software; you can redistribute it and/or modify it under
-the terms of the GNU Lesser General Public License as published by the
-Free Software Foundation; either version 3 of the License, or (at your
-option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
-
-This library is distributed in the hope that it will be useful, but WITHOUT
-ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
-FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
-more details.
-
-You should have received a copy of the GNU Lesser General Public License
-along with this library; if not, write to the Free Software Foundation, Inc.,
-51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
-**********/
-// "liveMedia"
-// Copyright (c) 1996-2023 Live Networks, Inc.  All rights reserved.
-// A sink that generates a QuickTime file from a composite media session
-// C++ header
-
-#ifndef _QUICKTIME_FILE_SINK_HH
-#define _QUICKTIME_FILE_SINK_HH
-
-#ifndef _MEDIA_SESSION_HH
-#include "MediaSession.hh"
-#endif
-
-class LIVEMEDIA_API QuickTimeFileSink: public Medium {
-public:
-  static QuickTimeFileSink* createNew(UsageEnvironment& env,
-				      MediaSession& inputSession,
-				      char const* outputFileName,
-				      unsigned bufferSize = 20000,
-				      unsigned short movieWidth = 240,
-				      unsigned short movieHeight = 180,
-				      unsigned movieFPS = 15,
-				      Boolean packetLossCompensate = False,
-				      Boolean syncStreams = False,
-				      Boolean generateHintTracks = False,
-				      Boolean generateMP4Format = False);
-
-  typedef void (afterPlayingFunc)(void* clientData);
-  Boolean startPlaying(afterPlayingFunc* afterFunc,
-                       void* afterClientData);
-
-  unsigned numActiveSubsessions() const { return fNumSubsessions; }
-
-protected:
-  QuickTimeFileSink(UsageEnvironment& env, MediaSession& inputSession,
-		    char const* outputFileName, unsigned bufferSize,
-		    unsigned short movieWidth, unsigned short movieHeight,
-		    unsigned movieFPS, Boolean packetLossCompensate,
-		    Boolean syncStreams, Boolean generateHintTracks,
-		    Boolean generateMP4Format);
-      // called only by createNew()
-  virtual ~QuickTimeFileSink();
-
-  virtual void noteRecordedFrame(MediaSubsession& inputSubsession,
-				 unsigned packetDataSize, struct timeval const& presentationTime);
-
-private:
-  Boolean continuePlaying();
-  static void afterGettingFrame(void* clientData, unsigned frameSize,
-				unsigned numTruncatedBytes,
-				struct timeval presentationTime,
-				unsigned durationInMicroseconds);
-  static void onSourceClosure(void* clientData);
-  void onSourceClosure1();
-  static void onRTCPBye(void* clientData);
-  void completeOutputFile();
-
-private:
-  friend class SubsessionIOState;
-  MediaSession& fInputSession;
-  FILE* fOutFid;
-  unsigned fBufferSize;
-  Boolean fPacketLossCompensate;
-  Boolean fSyncStreams, fGenerateMP4Format;
-  struct timeval fNewestSyncTime, fFirstDataTime;
-  Boolean fAreCurrentlyBeingPlayed;
-  afterPlayingFunc* fAfterFunc;
-  void* fAfterClientData;
-  unsigned fAppleCreationTime;
-  unsigned fLargestRTPtimestampFrequency;
-  unsigned fNumSubsessions, fNumSyncedSubsessions;
-  struct timeval fStartTime;
-  Boolean fHaveCompletedOutputFile;
-
-private:
-  ///// Definitions specific to the QuickTime file format:
-
-  unsigned addWord64(u_int64_t word);
-  unsigned addWord(unsigned word);
-  unsigned addHalfWord(unsigned short halfWord);
-  unsigned addByte(unsigned char byte) {
-    putc(byte, fOutFid);
-    return 1;
-  }
-  unsigned addZeroWords(unsigned numWords);
-  unsigned add4ByteString(char const* str);
-  unsigned addArbitraryString(char const* str,
-			      Boolean oneByteLength = True);
-  unsigned addAtomHeader(char const* atomName);
-  unsigned addAtomHeader64(char const* atomName);
-      // strlen(atomName) must be 4
-  void setWord(int64_t filePosn, unsigned size);
-  void setWord64(int64_t filePosn, u_int64_t size);
-
-  unsigned movieTimeScale() const {return fLargestRTPtimestampFrequency;}
-
-  // Define member functions for outputting various types of atom:
-#define _atom(name) unsigned addAtom_##name()
-  _atom(ftyp); // for MP4 format files
-  _atom(moov);
-      _atom(mvhd);
-  _atom(iods); // for MP4 format files
-      _atom(trak);
-          _atom(tkhd);
-          _atom(edts);
-              _atom(elst);
-          _atom(tref);
-              _atom(hint);
-          _atom(mdia);
-              _atom(mdhd);
-              _atom(hdlr);
-              _atom(minf);
-                  _atom(smhd);
-                  _atom(vmhd);
-                  _atom(gmhd);
-                      _atom(gmin);
-                  unsigned addAtom_hdlr2();
-                  _atom(dinf);
-                      _atom(dref);
-                          _atom(alis);
-                  _atom(stbl);
-                      _atom(stsd);
-                          unsigned addAtom_genericMedia();
-                          unsigned addAtom_soundMediaGeneral();
-                          _atom(ulaw);
-                          _atom(alaw);
-                          _atom(Qclp);
-                              _atom(wave);
-                                  _atom(frma);
-                                  _atom(Fclp);
-                                  _atom(Hclp);
-                          _atom(mp4a);
-//                            _atom(wave);
-//                                _atom(frma);
-                                  _atom(esds);
-                                  _atom(srcq);
-                          _atom(h263);
-                          _atom(avc1);
-                              _atom(avcC);
-                          _atom(mp4v);
-                          _atom(rtp);
-                              _atom(tims);
-                      _atom(stts);
-                      _atom(stss);
-                      _atom(stsc);
-                      _atom(stsz);
-                      _atom(co64);
-          _atom(udta);
-              _atom(name);
-              _atom(hnti);
-                  _atom(sdp);
-              _atom(hinf);
-                  _atom(totl);
-                  _atom(npck);
-                  _atom(tpay);
-                  _atom(trpy);
-                  _atom(nump);
-                  _atom(tpyl);
-                  _atom(dmed);
-                  _atom(dimm);
-                  _atom(drep);
-                  _atom(tmin);
-                  _atom(tmax);
-                  _atom(pmax);
-                  _atom(dmax);
-                  _atom(payt);
-  unsigned addAtom_dummy();
-
-private:
-  unsigned short fMovieWidth, fMovieHeight;
-  unsigned fMovieFPS;
-  int64_t fMDATposition;
-  int64_t fMVHD_durationPosn;
-  unsigned fMaxTrackDurationM; // in movie time units
-  class SubsessionIOState* fCurrentIOState;
-};
-
-#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/QuickTimeGenericRTPSource.hh

@@ -1,68 +0,0 @@
-/**********
-This library is free software; you can redistribute it and/or modify it under
-the terms of the GNU Lesser General Public License as published by the
-Free Software Foundation; either version 3 of the License, or (at your
-option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
-
-This library is distributed in the hope that it will be useful, but WITHOUT
-ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
-FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
-more details.
-
-You should have received a copy of the GNU Lesser General Public License
-along with this library; if not, write to the Free Software Foundation, Inc.,
-51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
-**********/
-// "liveMedia"
-// Copyright (c) 1996-2023 Live Networks, Inc.  All rights reserved.
-// RTP Sources containing generic QuickTime stream data, as defined in
-//     <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
-// C++ header
-
-#ifndef _QUICKTIME_GENERIC_RTP_SOURCE_HH
-#define _QUICKTIME_GENERIC_RTP_SOURCE_HH
-
-#ifndef _MULTI_FRAMED_RTP_SOURCE_HH
-#include "MultiFramedRTPSource.hh"
-#endif
-
-class LIVEMEDIA_API QuickTimeGenericRTPSource: public MultiFramedRTPSource {
-public:
-  static QuickTimeGenericRTPSource*
-  createNew(UsageEnvironment& env, Groupsock* RTPgs,
-	    unsigned char rtpPayloadFormat, unsigned rtpTimestampFrequency,
-	    char const* mimeTypeString);
-
-  // QuickTime-specific information, set from the QuickTime header
-  // in each packet.  This, along with the data following the header,
-  // is used by receivers.
-  struct QTState {
-    char PCK;
-    unsigned timescale;
-    char* sdAtom;
-    unsigned sdAtomSize;
-    unsigned short width, height;
-    // later add other state as needed #####
-  } qtState;
-
-protected:
-  virtual ~QuickTimeGenericRTPSource();
-
-private:
-  QuickTimeGenericRTPSource(UsageEnvironment& env, Groupsock* RTPgs,
-			    unsigned char rtpPayloadFormat,
-			    unsigned rtpTimestampFrequency,
-			    char const* mimeTypeString);
-      // called only by createNew()
-
-private:
-  // redefined virtual functions:
-  virtual Boolean processSpecialHeader(BufferedPacket* packet,
-                                       unsigned& resultSpecialHeaderSize);
-  virtual char const* MIMEtype() const;
-
-private:
-  char const* fMIMEtypeString;
-};
-
-#endif